Coding of stereo signals

ABSTRACT

A method of encoding a multi-channel signal having first and second signal components includes determining a set of filter parameters a prediction filter such that the prediction filter provides an estimate of the second signal component when receiving the first signal component as an input. The multi-channel signal is represented as the first signal component and the set of filter parameters. A corresponding decoding method and arrangements for encoding and decoding multi-channel signals are also provided.

This invention relates to the coding of multichannel signals includingat least a first and a second signal component. More particularly, theinvention relates to the coding of multiphonic audio signals, such asstereophonic signals.

Stereophonic audio signals comprise a left (L) and a right (R) signalcomponent which may originate from a stereo signal source, for examplefrom separated microphones. The coding of audio signals aims at reducingthe bit rate of a stereophonic signal, e.g. in order to allow anefficient transmission of sound signals via a communications network,such as the Internet, via a modem and analogue telephone lines, mobilecommunication channels or other a wireless networks, etc., and to storea stereophonic sound signal on a chip card or another storage mediumwith limited storage capacity.

U.S. Pat. No. 6,121,904 discloses a compressor for compressing digitalaudio signals comprising corresponding predictors for the left and rightstereo channels. The predictor for the left channel receives a currentsample and previous samples of the left audio signal as well as thecurrent and previous samples of the right audio signal and produces apredicted next sample of the left signal. Similarly, the predictor forthe right channel receives a current sample and previous samples of theright audio signal as well as the current and previous samples of theleft audio signal and produces a predicted next sample of the rightsignal.

It is an object of the present invention to provide a method of and anarrangement for coding multichannel signals with a low bit rate.

The above and other objects are achieved by a method of encoding amultichannel signal including at least a first signal component and asecond signal component, the method comprising the steps of

-   -   determining a set of filter parameters of a prediction filter        such that the prediction filter provides an estimate of the        second signal component when receiving the first signal        component as an input; and    -   representing the multichannel signal as the first signal        component and the set of filter parameters.

Consequently, by encoding the multichannel signal as a first signalcomponent and a set of filter parameters, the multichannel signal isencoded with a bit rate which is only slightly higher than that of asingle channel, e.g. a mono channel. The resulting encoded signal may bestored and/or communicated to a receiver. The invention is based on therecognition that for many multichannel signals one signal component maybe predicted from at least one other channel of the multichannel signalby an adaptive filter process. Consequently, when the determined filterparameters are communicated to a decoder, the multichannel signal may beretrieved on the basis of the first signal component and the filterparameters, allowing the decoder to model the second signal component.

The term multichannel signal comprises any signal including two or moreinterrelated signal components. Examples of such signals includemultiphonic audio signals, such as stereophonic signals, or the like,comprising synchronised recordings of the same audio presentation.According to some embodiments of the invention the multichannel signalcomprises transformed signal components of a multichannel source signal,e.g. transformed stereophonic signal components generated bytransforming the L and R stereo signals into a transformed set ofsignals which may be better suited for the modelling of one signalcomponent by another according to the invention. Further examples ofmulti-channel signals include signals received from a Digital VersatileDisc (DVD) or a Super Audio Compact Disc, etc.

In a preferred embodiment of the invention, the step of determining theset of filter parameters comprises the step of determining the filterparameters such that a difference of the second signal component and theestimated signal component is smaller than a predetermined value. Whenthe difference between the modelled signal and the second signalcomponent is small, the modelled signal provides a good estimate of thesecond signal component. Hence, a measure of quality is provided for themodelling of the second signal component, thereby ensuring that thecoding process according to the invention provides a minimum reductionin quality, e.g. in the example of stereo audio signals minimum audibledistortions of the signal.

According to a further preferred embodiment of the invention, the stepof representing the multichannel signal as the first signal componentand the set of filter parameters further comprises the step ofrepresenting the multichannel signal as the first signal component, theset of filter parameters, and an error signal indicative of thedifference of the second signal component and the estimated signalcomponent, if said difference is not smaller than said predeterminedvalue.

Hence, if the estimated signal provided by the step of filtering doesnot model the second signal component sufficiently well, the errorsignal is included in the encoded signal, thereby providing the decoderwith additional information. The decoder may combine the predictedsignal with the received error signal, thereby achieving a goodapproximation of the second signal component. The bit rate used forcommunicating the error signal may be varied, e.g. according to thebandwidth available for a communication link at a given time. Hence, itis an advantage of the invention that it provides the possibility for atrade-off between the bit rate used for communicating the signal and thesignal quality at the receiver. Therefore, a mechanism for gracefuldegradation is provided, e.g. by adaptively increasing or decreasing thebit rate allowed for the error signal.

In another preferred embodiment of the invention, the method furthercomprises the step of transforming at least a first source signalcomponent and a second source signal component of a multichannel sourcesignal into the first and second signal components. Consequently thefirst and second signal components are respective combinations of thefirst and second source signal components, thereby providing an inputsignal to the prediction filter which may be better suited forpredicting the second signal component as the corresponding sourcesignals. Examples of transformations include linear combinations of thefirst and second source signals, for example, in the case ofstereophonic audio signals the combinations L+R and L−R. Furtherexamples include rotations in signal space and other transformations.The transformation may be parameterised by transformation parameterswhich may be fixed or adaptive. i.e. they may be adapted according toproperties of the source signal.

In a further preferred embodiment of the invention,

-   -   said first signal component is a principal component signal of a        source multichannel signal including a number of source signal        components and the second signal component is a corresponding        residual signal;    -   the method further comprises the step of transforming at least        the first and second source signal components by a predetermined        transformation into the principal component signal including        most of the signal energy and at least the residual signal        including less energy than the principal component signal, the        predetermined transformation being parameterised by at least one        transformation parameter; and    -   the step of representing the multichannel signal as the first        signal component and the set of filter parameters further        comprises the step of representing the multichannel signal as        the principal component signal, the set of filter parameters,        and the transformation parameter.

Hence, according to this embodiment, the multichannel signal isrepresented by the principal signal, the transformation parameter, andthe set of filter parameters allowing the receiver to model the smallresidual signal, thereby improving the coding efficiency for themultichannel signal. This embodiment is based on the recognition thatfor many multichannel signals, e.g. in the case of audio signals formusic and speech signals, the residual signal may accurately beestimated as a filtered version of the principal signal. It is thereforean advantage of this embodiment that it provides a particularlyefficient method of encoding which preserves a high level of quality.

Preferably, the optimal transformation parameter may continuously betracked, thereby ensuring the transformation remains optimal even if thecharacteristics of the input signal changes, e.g. in the example of anaudio signal due to a moving sound source or changes in acousticproperties of the environment.

When the predetermined transformation is a rotation and thetransformation parameter corresponds to an angle of rotation, a simpletransformation is provided based only on a single parameter, the angleof rotation. By adapting the angle such that the signal components, e.g.the L and R signal components of a stereo signal, are rotated into aprincipal component signal and a residual signal, an efficient coding isprovided while maintaining a high quality signal.

It is an advantage of the invention that it provides an efficientbit-rate utilisation, i.e. a coding scheme which uses a low bit rate fora given sound quality. The coding scheme according to the invention maybe used to reduce the bit rate without significantly reducing the soundquality, to maintain the bit rate while improving the sound quality, ora combination of the above.

In a preferred embodiment of the invention, the step of determining aset of filter parameters further comprises the step of determining atleast one scaling parameter (β₁,β₂) for scaling the estimate of thesecond signal component such that a measure of correlation between thesecond signal component and the estimate of the second signal componentis increased. Consequently, a measure of similarity between theestimated and the actual signal is optimised, thereby further improvingthe quality of the coded signal.

The invention further relates to a method of decoding multichannelsignal information, the method comprising the steps of

-   -   receiving a first signal component and a set of filter        parameters;    -   estimating a second signal component using a prediction filter        corresponding to the received set of filter parameters, the        prediction filter receiving the received first signal component        as an input.

The present invention can be implemented in different ways including themethods described above and in the following, arrangements for encodingand decoding multichannel signals, respectively, a data signal, andfurther product means, each yielding one or more of the benefits andadvantages described in connection with the first-mentioned method, andeach having one or more preferred embodiments corresponding to thepreferred embodiments described in connection with the first-mentionedmethod and disclosed in the dependant claims.

It is noted that the features of the methods described above and in thefollowing may be implemented in software and carried out in a dataprocessing system or other processing means caused by the execution ofcomputer-executable instructions. The instructions may be program codemeans loaded in a memory, such as a RAM, from a storage medium or fromanother computer via a computer network. Alternatively, the describedfeatures may be implemented by hardwired circuitry instead of softwareor in combination with software.

The invention further relates to an arrangement for encoding amultichannel signal including at least a first signal component and asecond signal component the arrangement comprising

-   -   a prediction filter for estimating the second signal component,        the prediction filter corresponding to a set of filter        parameters and receiving the first signal component as an input;        and    -   processing means for representing the multichannel signal as the        first signal component and the set of filter parameters.

The invention further relates to an arrangement for decoding amultichannel signal corresponding to at least two signal components, thearrangement comprising

-   -   receiving means for receiving a first signal component of the        multichannel signal and a set of filter parameters;    -   a prediction filter for estimating a second signal component of        the multichannel signal, the prediction filter receiving the        received set of filter parameters and the received first signal        component as an input.

The above arrangements may be part of any electronic equipment includingcomputers, such as stationary and portable PCs, stationary and portableradio communications equipment and other handheld or portable devices,such as mobile telephones, pagers, audio players, multimedia players,communicators, i.e. electronic organisers, smart phones, personaldigital assistants (PDAs), handheld computers, or the like.

The term processing means comprises general- or special-purposeprogrammable microprocessors, Digital Signal Processors (DSP),Application Specific Integrated Circuits (ASIC), Programmable LogicArrays (PLA), Field Programmable Gate Arrays (FPGA), special purposeelectronic circuits, etc., or a combination thereof. The above first andsecond processing means may be separate processing means or they may becomprised in one processing means.

The term receiving means includes circuitry and/or devices suitable forenabling the communication of data, e.g. via a wired or a wireless datalink. Examples of such receiving means include a network interface, anetwork card, a radio receiver, a receiver for other suitableelectromagnetic signals, such as infrared light, e.g. via an IrDa port,radio-based communications, e.g. via Bluetooth transceivers, or thelike. Further examples of such receiving means include a cable modem, atelephone modem, an Integrated Services Digital Network (ISDN) adapter,a Digital Subscriber Line (DSL) adapter, a satellite transceiver, anEthernet adapter, or the like.

The term receiving means further comprises other input circuits/devicesfor receiving data signals, e.g. data signals stored on acomputer-readable medium. Examples of such receiving means include afloppy-disk drive, a CD-Rom drive, a DVD drive, or any other suitabledisc drive, a memory card adapter, a smart card adapter, etc.

The invention further relates to a data signal including multichannelsignal information, the data signal being generated by a methoddescribed above and in the following. The signal may be embodied as adata signal on a carrier wave, e.g. as a data signal transmitted bycommunications means as described above and in the following.

The invention further relates to a computer-readable medium comprising adata record indicative of multichannel signal information generated by amethod described above and in the following. The term computer-readablemedium comprises magnetic tape, optical disc, digital video disk (DVD),compact disc (CD or CD-ROM), mini-disc, hard disk, floppy disk,ferro-electric memory, electrically erasable programmable read onlymemory (EEPROM), flash memory, EPROM, read only memory (ROM), staticrandom access memory (SRAM), dynamic random access memory (DRAM),synchronous dynamic random access memory (SDRAM), ferromagnetic memory,optical storage, charge coupled devices, smart cards, PCMCIA card, etc.

The invention further relates to a device for communicating amultichannel signal, the device comprising an arrangement for encodingthe multichannel signal as described above and in the following.

These and other aspects of the invention will be apparent from andelucidated with reference to the embodiments and with reference to thedrawing, in which:

FIG. 1 shows a schematic view of a system for communicating stereosignals according to an embodiment of the invention;

FIG. 2 shows a schematic view of an arrangement for encoding amultichannel signal according to a first embodiment of the invention;

FIG. 3 shows a schematic view of an arrangement for decoding amultichannel signal according to the first embodiment of the invention;

FIG. 4 shows a schematic view of an arrangement for encoding a stereosignal according to a second embodiment of the invention;

FIG. 5 illustrates the determination of the signal transformationaccording to an embodiment of the invention;

FIG. 6 shows a schematic view of an arrangement for decoding a stereosignal according to the second embodiment of the invention;

FIGS. 7 a-c show schematic views of examples of a filter circuit for usein an embodiment of the invention;

FIG. 8 shows a Schematic view of an arrangement for encoding a stereosignal according to a third embodiment of the invention;

FIG. 9 shows a schematic view of an arrangement for encoding a stereosignal according to a fourth embodiment of the invention;

FIG. 10 shows a schematic view of an arrangement for decoding a stereosignal according to the fourth embodiment of the invention;

FIG. 11 shows a schematic view of an arrangement for encoding amultichannel signal according to a fifth embodiment of the invention;and

FIG. 12 shows a schematic view of a subtraction circuit for use with anembodiment of the invention.

FIG. 1 shows a schematic view of a system for communicating stereosignals according to an embodiment of the invention. The systemcomprises a coding device 101 for generating a coded stereophonic signaland a decoding device 105 for decoding a received coded signal into astereo L signal and a stereo R signal component. The coding device 101and the decoding device 105 each may be any electronic equipment or partof such equipment. Here the term electronic equipment comprisescomputers, such as stationary and portable PCs, stationary and portableradio communication equipment and other handheld or portable devices,such as mobile telephones, pagers, audio players, multimedia players,communicators, i.e. electronic organisers, smart phones, personaldigital assistants (PDAs), handheld computers, or the like. It is notedthat the coding device 101 and the decoding device may be combined inone electronic equipment where stereophonic signals are stored on acomputer-readable medium for later reproduction.

The coding device 101 comprises an encoder 102 for encoding astereophonic signal according to the invention, the stereophonic signalincluding an L signal component and an R signal component. The encoderreceives the L and R signal components and generates a coded signal T.The stereophonic signal L and R, may originate from a set ofmicrophones, e.g. via further electronic equipment, such as a mixingequipment, etc. The signals may further be received as an output fromanother stereo player, over-the-air as a radio signal, or by any othersuitable means. Preferred embodiments of such an encoder according tothe invention will be described below. According to one embodiment, theencoder 102 is connected to a transmitter 103 for transmitting the codedsignal T via a communications channel 109 to the decoding device 105.The transmitter 103 may comprise circuitry suitable for enabling thecommunication of data, e.g. via a wired or a wireless data link 109.Examples of such a transmitter include a network interface, a networkcard, a radio transmitter, a transmitter for other suitableelectromagnetic signals, such as an LED for transmitting infrared light,e.g. via an IrDa port, radio-based communications, e.g. via a Bluetoothtransceiver, or the like. Further examples of suitable transmittersinclude a cable modem, a telephone modem, an Integrated Services DigitalNetwork (ISDN) adapter, a Digital Subscriber Line (DSL) adapter, asatellite transceiver, an Ethernet adapter, or the like.Correspondingly, the communications channel 109 may be any suitablewired or wireless data link, for example of a packet-basedcommunications network, such as the Internet or another TCP/IP network,a short-range communications link, such as an infrared link, a Bluetoothconnection or another radio-based link. Further examples of thecommunications channel include computer networks and wirelesstelecommunications networks, such as a Cellular Digital Packet Data(CDPD) network, a Global System for Mobile (GSM) network, a CodeDivision Multiple Access (CDMA) network, a Time Division Multiple AccessNetwork (TDMA), a General Packet Radio service (GPRS) network, a ThirdGeneration network, such as a UMTS network, or the like. Alternativelyor additionally, the coding device may comprise one or more otherinterfaces 104 for communicating the coded stereo signal T to thedecoding device 105. Examples of such interfaces include a disc drivefor storing data on a computer-readable medium 110, e.g. a floppy-diskdrive, a read/write CD-ROM drive, a DVD-drive, etc. Other examplesinclude a memory card slot a magnetic card reader/writer, an interfacefor accessing a smart card, etc. Correspondingly, the decoding device105 comprises a corresponding receiver 108 for receiving the signaltransmitted by the transmitter and/or another interface 106 forreceiving the coded stereo signal communicated via the interface 104 andthe computer-readable medium 110. The decoding device further comprisesa decoder 107 which receives the received signal T and decodes it intocorresponding stereo components L′ and R′. Preferred embodiments of sucha decoder according to the invention will be described below. Thedecoded signals L′ and R′ may subsequently be fed into a stereo playerfor reproduction via a set of speakers, head-phones, or the like.

FIG. 2 shows a schematic view of an arrangement for encoding amultichannel signal according to a first embodiment of the invention.According to this embodiment, the multichannel signal comprises twocomponents S₁ and S₂. The arrangement comprises an adaptive filter 201receiving the signal component S₁ as an input and generating a filteredsignal Ŝ₂. The filter parameters F_(p) of the adaptive filter areselected such that the filtered signal Ŝ₂ approximates the second signalcomponent S₂, e.g. by controlling the adaptive filter 201 by the errorsignal e indicating the difference between S₂ and Ŝ₂ as generated by asubtraction circuit 203. The filter 201 may be any suitable filter knownin the art. Examples of such filters include a finite impulse response(FIR) filter or a infinite impulse response (IIR) filter, adaptive orfixed, with the cut-off frequencies and magnitudes being fixed ortracked recursively, or the like. The filter may be of any order,preferably smaller than 10. The type of the filter can be Butterworth,Chebychev, or any other suitable type of filter. In the example of audiosignals, examples of such adaptive filters include an adaptive filterknown from the field of echo cancellation, or a filter based on apsychoacoustic model of the human auditory system, e.g. as is known fromMPEG coding, thereby reducing the number of filter parameters. Accordingto another embodiment the filter may further be simplified, e.g. byusing a 10^(th) order filter using 5 BiQuadratic filters and anartificial reverberation unit. In this embodiment, at the encoding side,the filter is fitted and the reverberation time is determined. Theseparameters are varying slowly, thereby reducing the necessary bit ratefor their transmission.

The resulting filter parameters F_(p) are fed into an encoder 205, e.g.an encoder providing a Huffman encoding or any other suitable codingscheme, resulting in encoded filter parameters F_(pe). The encodedfilter parameters F_(pe) are fed into a combiner circuit 204. Thearrangement further comprises encoders 202 performing a proper encodingof the signal component S₁. For example, in the case of audio signals,the signal S₁ may be encoded according to MPEG, e.g. MPEG I layer 3(MP3), according to sinusoidal coding (SSC), or audio coding schemesbased on subband, parametric, or transform schemes, or any othersuitable schemes or combination thereof. The resulting coded signalS_(1,e) is fed into the combiner circuit 204 together with the filterparameters F_(p). The combiner circuit 204 performs framing, bit-rateallocation, and lossless coding, resulting in a combined signal T to becommunicated.

FIG. 3 shows a schematic view of an arrangement for decoding amultichannel signal according to the first embodiment of the invention.The arrangement receives a coded multichannel signal T, for exampleoriginating from an encoder according to the embodiment described inconnection of FIG. 2. The arrangement comprises a circuit 301 forextracting the encoded signal S_(1,e) and the encoded filter parametersF_(pe) from the combined signal T, i.e. the circuit 301 performs aninverse operation of the combiner 204 of FIG. 2. The filter parametersare decoded by a decoder 303 corresponding to the encoding of the filterparameters by the encoder 205 of FIG. 2. The extracted signal S_(1,e) isfed into a decoder 302 for performing audio decoding corresponding tothe encoding performed by the encoder 202 of FIG. 2, resulting in thedecoded first signal component signal S₁′. The signal S₁′ is fed into afilter 304 together with the decoded filter parameters F_(p). The filter304 generates a corresponding estimated second signal component Ŝ₂′.Hence, the decoder of FIG. 2 generates an output corresponding to thereceived first signal component S₁′ and the estimated second signalcomponent Ŝ₂′.

FIG. 4 shows a schematic view of an arrangement 102 for encoding astereo signal according to a second embodiment of the invention. Thearrangement comprises circuitry 401 for performing a rotation of thestereo signal in the L-R space by an angle α, resulting in rotatedsignal components y and r according to the transformationy=L cos α+R sin αw _(L) L+w _(R) Rr=−L sin α+R cos α=−w _(R) L+w _(L) R,  (1)where w_(L)−cosα and w_(R)=sinα will be referred to as weightingfactors.

According to this embodiment, the angle α is determined such that itcorresponds to a direction of high signal variance. The direction ofmaximum signal variance, i.e. the principal component, may be estimatedby a principal component analysis such that the rotated y componentcorresponds to the principal component signal which includes most of thesignal energy, and r is a residual signal. Correspondingly, thearrangement of FIG. 4 comprises circuitry 400 which determines the angleα or, alternatively, the weight factors w_(L) and w_(R).

Referring to FIG. 5, according to a preferred embodiment, the aboveweight factors w_(L) and w_(R) are determined according the followingalgorithm:

Initially, the incoming stereo signals L and R are rectified and lowpassfiltered, resulting in envelope signals p(k) of L and q(k) of R,respectively, where p(k) and q(k) are suitably sampled and the sampleindex is denoted k. Thus, the vector x(k)=(p(k), q(k)) denotes theincoming signal vector. Alternatively, the signals L and R may be useddirectly, i.e. without filtering, or other filtered versions of L and Rmay be used, e.g. highpass filtered signals L and R. In FIG. 5 a numberof signal points are illustrated as circles. As an example, the signalpoint x(k) and its corresponding components p(k) and q(k) are indicated.According to the invention, the signals are rotated in the direction ofthe principal component of the signal vectors. In the example of FIG. 5,this corresponds to the y direction where a is the angle between the ydirection and the p direction. The weight vector w=(w_(L), w_(R))indicates the direction of the principal component, and the rotatedcomponents of x(k) are denoted y(k) and r(k), respectively.

The principal component may be determined by any suitable method knownin the art. In a particularly advantageous embodiment, an iterativemethod utilising Oja's rule (see e.g. S. Haykin: “Neural Networks”,Prentice Hall, N.J., 1999) is used. According to this embodiment, theweight vector w is iteratively estimated according to the followingequationw(k)=w(k−1)+μ[x(k−1)−w(k−1) y(k−1)],  (2)where w(k)=(w_(L)(k), w_(R)(k)) corresponds to the estimate at time k.The above iteration may, for example, be initiated with a set of smallrandom weights w(0), or in any other suitable way. The above estimatedweight vector may be used to calculate the rotated signal according toy(k)=w^(T)(k)x(k). Alternatively, the iteration of eqn. (2) may beperformed on a block basis, e.g. for a block of N samples, where Ndepends on the particular implementation, for example, N=512, 1024,2048, etc. In this embodiment, the estimated weight vector w(N) for ablock may be used in the transformation of all samples of that blockaccording to y(k)=w^(T)(N)x(k).

The factor μin eqn. (2) corresponds to a time scale of the trackingalgorithm. If μ=0, the weighting factors and, thus, the angle α, remainconstant, while they change rapidly for large μ. As an example, for ablock size of 2048 samples, μ may be selected of the order of 10⁻³ for asampling rate of 44.1 kHz.

It is an advantage of the above iterative algorithm that it is linear,i.e. it does not require the calculation of any trigonometric functions,square roots or the like. It is a further advantage, that the aboveiteration yields a normalised weight vector w, as the term—μw(k−1)y(k−1) in eqn. (2) corresponds to a weight decay term penalisinglarge weights while the term +μx(k−1) drives the weight vector in thedirection of the principal component. It is further noted that in thepresent embodiment, since x(k) is the envelope signal, w_(L), w_(R) ∈[0,1], i.e. the weight vector w lies in the first quadrant in FIG. 5,thereby ensuring that μ is positive. It is a further advantage of thisembodiment that it suffices to transmit one of w_(L) and w_(R), as theother factor may be determined according to w_(R)=√{square root over(1−(w_(L))²)}. Alternatively, the angle α may be transmitted.

Again referring to FIG. 4, the circuit 400 outputs the determined angleαor, alternatively, one or both of the weight factors w_(L) and w_(R).The angle information is fed into the rotation circuit 401 whichgenerates the rotated signal components y and r. It is understood thatthe circuits 400 and 401 may be combined in a single circuit performingthe iterative calculation of eqn. (2) and the calculation of y and raccording to eqn. (1).

According to this embodiment of the invention, it is recognised that theresidual signal r may be estimated as a filtered version of theprincipal signal y. In an acoustic recording of an audio source recordedby two microphones in the absence of acoustic distortions, e.g. due toreflections, etc., the principal signal y corresponds to the audiosource and the residual signal is substantially zero. For example, thestereo signals L and R may be expressed as L=M+S and R=M−S, where Mcorresponds to a mid or centre signal and S corresponds to a stereo orside signal. In the case of an acoustic recording of a stationary soundsource, e.g. a speaker recorded by two microphones, the L and R signalsare substantially equal, if the speaker is positioned exactly betweenthe microphones and assuming that there are no acoustic distortions suchas reflections, etc. Hence, in this case S is substantially zero or atleast small and the coding scheme according to this embodimentsubstantially yields y corresponding to L+R and r corresponding to L−Rbeing zero or small; this corresponds to α=45 degrees. If the speaker isnot positioned exactly between the microphones, i.e. there is anasymmetry, but still assuming that there are no reflections or otherdistortions, the rotated signal y according to the invention stillcorresponds to the speaker and the residual signal r is substantiallyzero. However, in this case the angle α differs from 45 degrees.

In a more realistic situation distortions are present, e.g. due toreflections of the signal at the walls of a room and at the head andtorso of the speaker, etc. These effects influence the residual signalr. Consequently, when estimating the residual signal by a filter, thefilter in effect models the room acoustics, etc. For a classicalorchestra the situation is similar, while in the case of modem pop musicthe situation may be slightly different. In this case, a sound engineertypically mixes multiple channels into two channels, often usingartificial reverberation, effect boxes etc. In this case the filtermodels the acoustic effects introduced by the mixing process.

Accordingly, still referring to FIG. 4, the arrangement furthercomprises an adaptive filter 201 receiving the principal signal y as aninput and generating a filtered signal . The filter parameters F_(p) ofthe adaptive filter are selected such that the filtered signalapproximates the residual signal r, e.g. by controlling the adaptivefilter 201 by the error signal e indicating the difference between and{circumflex over (r)}^ as generated by a subtraction circuit 402. Theresulting filter parameters F_(p) are fed into an encoder 205, e.g. anencoder providing a Huffman encoding or any other suitable codingscheme, resulting in encoded filter parameters F_(pe). The encodedfilter parameters F_(pe)are fed into a combiner circuit 204. The filter201 may be any suitable filter known in the art. Example of such filtersinclude a finite impulse response (FIR) filter or a infinite impulseresponse (TIE) filter, adaptive or fixed, with the cut-off frequenciesand magnitudes being fixed or tracked recursively, or the like. Thefilter may be of any order, preferably smaller than 10. The type of thefilter can be Butterworth, Chebychev, or any other suitable type offilter. The arrangement further comprises an encoder 202 for encodingthe principal signal as described in connection with FIG. 2, resultingin the encoded principal signal Y_(e) which is fed into the combinercircuit 204 together with the filter parameters F_(p) and the angleinformation a. As described in connection with FIG. 2, the combinercircuit 204 performs framing, bit-rate allocation, and lossless coding,resulting in a combined signal T to be communicated which includes theencoded principal signal Y_(e), the filter parameters F_(p) and theangle information α. In one embodiment, the angle α or, alternatively,W_(L) and/or W_(R) may be communicated as part of a header transmittedprior to a signal frame, a signal block, or the like.

According to the invention, as the transformation angle α is trackedsuch that the principal component signal includes most of the signalenergy, the bit rates allocated to the y and r signals may be selectedto be different, thereby optimising the coding efficiency. As describedabove, in the example of an acoustic recording of an audio sourcerecorded by two microphones in the absence of acoustic distortions, theprincipal signal y corresponds to the audio source and the residualsignal is substantially zero. In this example, the angle α correspondsto the position of the sound source relative to the microphones. If thesound source moves, e.g. from left to right, the method according to theinvention still yields a principal component signal y corresponding tothe source and a small residual signal r, ideally being r=0. In thiscase, a changes form 0 (fully left) to 90 degrees (fully right). Theabove example illustrates the advantage of tracking the angle α. Hence,it is an advantage of the invention that it allows an efficient codingof stereo signals.

According to this embodiment of the invention, the bit rate to beallocated to the filter parameters F_(p) may be considerably smallerthan the bit rate necessary for the principal signal y, e.g. in oneembodiment, the bit-rate for F_(p) may, on average, be less than 10% ofthe bit rate for y. Hence, it is an advantage of the invention that itreduces the bit rate necessary for transmitting a stereo signal. Thetotal bit rate according to the invention is only slightly higher thanfor a single mono channel. It is noted, however, that this ratio mayvary during a recording. For example, the ratio may become smaller, e.g.in a situation with little distortions and a stationary source, but alsolarger, e.g. if the L and R signals are momentarily independent.

FIG. 6 shows a schematic view of an arrangement 107 for decoding astereo signal according to the second embodiment of the invention. Thearrangement receives a coded stereo signal T, for example originatingfrom an encoder according to the embodiment described in connection withFIG. 4. The arrangement comprises a circuit 301 for extracting theencoded signals y_(e), the encoded filter parameters F_(pe), and theangle information a from the combined signal T, i.e. the circuit 301performs an inverse operation of the combiner 204 of FIG. 4. Theextracted signal y_(e) is fed into a decoder 302 for performing audiodecoding corresponding to the encoding performed by the encoder 202 ofFIG. 4, resulting in the decoded principal component signal y′. Theencoded filter parameters F_(pe) are decoded by a decoder 303corresponding to the encoding of the filter parameters by the encoder205 of FIG. 4. The signal y′ is fed into a filter 304 together with thedecoded filter parameters F_(p). The filter 304 generates acorresponding estimated residual signal ρ′. The received principalcomponent signal y′, the estimated residual signal ρ′ and the receivedangle information a are fed into a rotation circuit 601 which rotatesthe signals y′, ρ′ back in the direction of the original L and Rcomponents, thus resulting in the received signals L′ and R′.

In the embodiment described in connection with FIGS. 4 and 6, thefilters 201 and 304 may be standard adaptive filters in the temporal ortime domain (see e.g. “Adaptive Filter Theory”, by S. Haykin, PrenticeHall, 2001), e.g. an adaptive filter known from the field of echocancellation. Other examples of filters include a fixed FIR or IIRfilter with a fixed or adaptive cut-off-frequency and magnitude.Alternatively, the filter may be based on a psychoacoustic model of thehuman auditory system or another suitable filter, e.g. using a 10^(th)order filter using 5 BiQuadratic filters and an artificial reverberationunit, as described in connection with FIG. 2.

FIGS. 7 a-c show schematic views of examples of a filter circuit for usein an embodiment of the invention.

In the example of FIG. 7 a, the filter 201 comprises a combination of afilter 701 and a reverberation filter 702. For example, the filter 701may be a standard adaptive filter in the temporal or time domain, afixed FIR or IIR filter with a fixed or adaptive cut-off-frequency andmagnitude, etc., e.g. a high-pass filter. According to this embodiment,both the filter parameters of the filter 701 and the parameters of thereverberation filter 702, such as the reverberation time denoted T₆₀,are transmitted to the decoder as filter parameters F_(p).

In the example of FIG. 7 b, in addition to the filters 701 and 702, twocontrol circuits 703-704 are added. A control circuit 703 is added toensure that the average power of the residual signal r and the averagepower of the output of the reverberator 702 are approximately the same,e.g. by multiplying the output of the reverberator 702 with a parameterβ₁. A second control circuit 704 multiplies the scaled output of thereverberator with β₂. The factor β₂ may be selected in the range between−3 dB and +6 dB and it is determined such that the cross correlation ρbetween r and ρ is as high as possible, i.e. that the signals r and ρare as similar as possible. Hence, the filter arrangement of FIG. 7 bfurther comprises a circuit 705 for determining the cross correlation ρ.The filter arrangement further comprises a multiplier 706 for generatingthe product β=β₁·β₂ which is output as a part of the filter parametersF_(p). Hence, β₁ is a gain that is automatically controlled, e.g. bycomparing the absolute mean of r and ρ, and β2 is another gain that isautomatically controlled, e.g. by use of the cross-correlationcoefficient ρ. The first gain is intended to make sure that the energyof r is preserved, i.e. that the energy of the predicted signal ρ′ atthe receiver corresponds to the energy of r. The second gain is to makesure that r and ρ′ are well correlated.

In one embodiment, the reverberator 702 and the filter 701 may be fixed,i.e. not adapted according to the filter parameters F_(p). Further, β2may be fixed, thereby leaving the slowly varying parameter β₁ as theonly adaptive parameter which needs to be adjusted and transmitted.Consequently, a particularly simple filter arrangement is provided. Itis an advantage of this embodiment that it only requires about half theoriginal stereo bit rate for transmitting a stereo signal. It is notedthat further variations of the above embodiment may be used. Forexample, in one embodiment the filter 701 may be left out.

Furthermore, alternatively or additionally to the correlation p, othermeasures of correlation may be used to ensure a high degree ofsimilarity between the original signal and the signal afterencoding-decoding. For example, in one embodiment two correlators may beused instead of correlator 705. One correlator may compute thecross-correlation ρ_(LR) of the input signals L and R. Furthermore, asecond correlator may compute the cross correlation ρ′_(LR) of theresulting outputs L′ and R′ of the encoder-decoder, i.e. according tothis embodiment, the encoder further comprises a decoder circuit fordetermining the signals L′ and R′. This embodiment uses the differenceε_(ρLR)=ρ_(LR)−ρ′_(LR) to control β₂ such that ερ is minimal. This isillustrated in FIG. 7 c, where the correlator of FIG. 7 b is replaced bycircuit 707 which receives the signals L and R as well as L′ and R′ asinputs and generates as an output a signal indicative of the differenceε_(p). The output ε_(p) of circuit 707 controls circuit 704 to scale theestimated residual ρ such that ε_(p) is minimised. In one embodiment,the inputs to circuit 707 are high-pass filtered, e.g. at 250 Hz, suchthat the low frequencies have a decreasing contribution to ε_(ρ). As inthe embodiment of FIG. 7 b, it is an advantage of this embodiment thatthe correlation between the resulting stereo image and the originalstereo image before the coding-decoding is very high.

FIG. 8 shows a schematic view of an arrangement for encoding a stereosignal according to a third embodiment of the invention. The arrangementis a variation of the embodiment described in connection with FIG. 4,and it comprises circuitry 401 for performing a rotation of the stereosignals L and R, circuitry 400 for determining the angle of rotation, anadaptive filter 201, a subtraction circuit 203, an encoder 202, anencoder 205, and a combiner circuit 204, as described in connection withFIG. 4. According to this embodiment, the principal component signal yis not directly fed into the filter 201. Instead, the arrangementfurther comprises a decoder 302 as described in connection with FIG. 6.The decoder 302 receives the encoded principal component signal y_(e)generated by the encoder 202 and generates the decoded principal signaly′ which is fed into the filter 201. It is an advantage of thisembodiment that it reduces the effect of coding errors introduced by thecoding and decoding of the signal y. These coding errors cause thedecoded signal y′ to be slightly different from the original signal ydue to the fact that the decoder 302 in practice is not a perfectinverse of the encoder 202, i.e. E E⁻¹≠1. Consequently, by applying anencoding and decoding of the signal y at the decoder, the input y′ tothe filter 201 corresponds to the input y′ fed into the filter 304 (ofFIG. 6) at the receiver, thereby improving the result of the predictionof ρ′ of the residual signal at the receiver. Hence, the encoderaccording to this embodiment may be used in connection with a decoderaccording to the embodiment of FIG. 6.

FIG. 9 shows a schematic view of an arrangement for encoding a stereosignal according to a fourth embodiment of the invention. Thearrangement is a variation of the embodiment described in connectionwith FIG. 4, and it comprises circuitry 401 for performing a rotation ofthe stereo signals L and R, circuitry 400 for determining the angle ofrotation, an adaptive filter 201, a subtraction circuit 203, an encoder202, an encoder 205, and a combiner circuit 204, as described inconnection with FIG. 4. According to this embodiment, the principalcomponent signal y is not directly fed into the filter 201. Instead, thearrangement further comprises a multiplication circuit 901 multiplyingthe residual signal r received from circuit 401 with a constant γ, andan adding circuit 902 for adding the scaled residual signal to theprincipal component signal y, resulting in a signal y+γr which is fedinto the filter 201. Here, γ is a small positive value, e.g. of theorder of 10⁻². In one embodiment, the constant γ is tracked adaptively.It is an advantage of this embodiment that frequencies which aresubstantially not present in the spectrum of the signal y but present inthe spectrum of r may be utilised in the modelling of the residualsignal ρ by the filter 201, thereby improving the quality of the codedsignal. According to this embodiment the signal y+γ r is fed into theencoder 202 which generates the decoded principal signal y_(e) to betransmitted to the receiver. Furthermore, according to this embodiment,the constant γ is fed into the combiner 204 and transmitted to thereceiver.

FIG. 10 shows a schematic view of an arrangement for decoding a stereosignal according to the fourth embodiment of the invention, i.e.suitable for decoding a signal received from an encoder according toFIG. 9. The arrangement comprises a circuit 301 for extracting thereceived information from the combined signal T, a decoder 302, adecoder 303, a filter 304, and a rotation circuit 601 as described inconnection with FIG. 6. According to this embodiment, the circuit 301further extracts the constant γ from the combined signal T, and thearrangement further comprises a multiplication circuit 1001 formultiplying the predicted residual signal ρ′ generated by the filter 304with the received constant γ. The arrangement further comprises acircuit 1002 for subtracting the resulting scaled predicted residualsignal γρ′ from the decoded principal signal γ′.

FIG. 11 shows a schematic view of an arrangement for encoding amulti-channel signal according to a fifth embodiment of the invention.The arrangement receives a multichannel signal comprising n channels S₁,. . . S_(n). The arrangement comprises a principal component analyser1100 for performing a principal component analysis of the signalcomponents S₁, . . . , S_(n), resulting in a weight vector w=(w₁, . . ., w_(n)) for transforming the input signal into a principal componentsignal y and n-1 residual signals r₁, r₂, . . . , r_(n-1). Thearrangement further comprises a transformation circuit 1101 receivingthe input signal components S₁, . . . , S_(n) and the determined weightvector w, and generating the signals y and r₁, . . . , r_(n-1) accordingto the above transformation. The principal component signal y is fedinto a set of adaptive filters 201, each predicting one of the residualsignals r₁, . . . , r_(n), as described in connection with FIG. 4,resulting in corresponding filter parameters F_(p1), . . . , F_(p(n-1))which are fed into corresponding encoders 205 and, subsequently, intothe combiner 204. At a corresponding decoder (not shown), correspondingfilters are used for generating estimates ρ′₁, . . . , ρ′_(n-1) of theresidual signals based on the filter parameters, as described inconnection with FIG. 6. The arrangement further comprises an encoder 202for encoding the principal component signal y, resulting in an encodedsignal y_(e) which is also fed into the combiner 204.

It is understood that, according to one embodiment, only a subset ofresidual signals, e.g. r₁, . . . , r_(k), k<n−1, may be transmitted tothe receiver or fed into corresponding filters, thereby reducing thenecessary bit rate while maintaining most of the signal quality.

FIG. 12 shows a schematic view of a subtraction circuit for use with anembodiment of the invention. In the above embodiments, the filterparameters are determined by comparing a target signal with an estimatedsignal, i.e. by the error signal e indicating the difference between rand ρ as generated by a subtraction circuit 203. It is understood thatthe subtraction circuit may generate different measures of differencebetween r and ρ, for example a difference may be determined in the timedomain or in the frequency domain. Referring to FIG. 12, the circuit 203may comprise circuits 1201 for transforming the signals r and ρ,respectively, into the frequency domain, e.g. by performing a fastFourier transformation (FFT). The resulting frequency components may befurther processed by respective circuits 1204. For example differentfrequencies may be weighted differently, preferably according to theproperties of the human auditory system, thereby weighting differencesin the audible frequency range more strongly. Other examples of furtherprocessing by the circuits 1204 include an averaging over predeterminedfrequency components, calculating the magnitude of the complex frequencycomponents, clustering of filter components, or the like. For example,in a preferred embodiment, a clustering is performed prior to thesubtraction in the frequency domain. This clustering may be performedusing a filter-bank, e.g. with linear or logarithmic sub-bandwidths.Alternatively, the clustering may be performed using the so-calledequivalent rectangular bandwidth (ERB) (see e.g. “An introduction to thePsychology of Hearing”, by Brian Moore, Academic Press, London, 1997).The equivalent rectangular bandwidth technique clusters frequency-bandsthat correspond to the human auditory filters, e.g. the so-calledcritical bands. According to this embodiment, the corresponding value ofthe ERB as a function of centre frequency, f (in kHz), is may becalculated according to ERB=24.7(4.37 f+1). Still referring to FIG. 12,the circuit 203 further comprises a subtraction circuit 1203 forsubtracting the processed frequency components. Alternatively, thetransformed signals generated by the circuits 1201 are directly fed intothe subtraction circuit 1204 without further processing. The differencesignal generated by the subtraction circuit 1204 is fed into atransformation circuit 1202 for transforming the error signal back intothe time domain, e.g. by performing an inverse fast Fourier transform(IFFT). Alternatively, the difference signal in the frequency domain maybe used directly.

It is understood that a skilled person may adapt the above embodiments,e.g. by adding or removing features, or by combining features of theabove embodiments. For example, it is understood that the featuresintroduced in embodiments of FIGS. 8 and 9 may be incorporated in theembodiment of FIG. 11 as well. As another example, the error signal edescribing the quality of the estimated residual signal in theembodiment of FIG. 4 may be compared to a threshold error indicating amaximum acceptable error. If the error is not acceptable, the errorsignal may, after suitable coding, be transmitted together with thesignal T similar to the methods used within the field of LinearPredictive Coding (LPC).

It is further noted that the invention is not limited to stereophonicsignals, but may also be applied to other multi-channel input signalshaving two or more input channels. Examples of such multi-channelsignals include signals received from a Digital Versatile Disc (DVD) ora Super Audio Compact Disc, etc. In this more general case, a principalcomponent signal y and one or more residual signals r may still begenerated according to the invention. The number of residual signalstransmitted depends on the number of channels and the desired bit rate,as higher order residuals may be omitted without significantly degradingthe signal quality.

In general, it is an advantage of the invention that bit-rate allocationmay be adaptively varied, thereby allowing graceful degradation. Forexample, if the communication channel momentarily only allows a reducedbit rate to be transmitted, e.g. due to increased network traffic,noise, or the like, the bit rate of the transmitted signal may bereduced without significantly degrading the perceptible quality of thesignal. For example, in the case of a stationary sound source discussedabove, the bit rate may be reduced by a factor of approximately twowithout significantly degrading the signal quality, corresponding totransmitting a single channel instead of two.

It is noted that the above arrangements may be implemented as general-or special-purpose programmable microprocessors, Digital SignalProcessors (DSP), Application Specific Integrated Circuits (ASIC),Programmable Logic Arrays (PLA), Field Programmable Gate Arrays (FPGA),special purpose electronic circuits, etc., or a combination thereof.

It should be noted that the above-mentioned embodiments illustraterather than limit the invention, and that those skilled in the art willbe able to design many alternative embodiments without departing fromthe scope of the appended claims. In the claims, any reference signsplaced between parentheses shall not be construed as limiting the claim.The word ‘comprising’ does not exclude the presence of other elements orsteps than those listed in a claim. The invention can be implemented bymeans of hardware comprising several distinct elements, and by means ofa suitably programmed computer. In a device claim enumerating severalmeans, several of these means can be embodied by one and the same itemof hardware. The mere fact that certain measures are recited in mutuallydifferent dependent claims does not indicate that a combination of thesemeasures cannot be used to advantage.

1. A method of encoding a multi-channel signal including at least afirst signal component and a second signal component, the first signalcomponent being a principal component signal of a multi-channel sourcesignal including a number of source signal components and the secondsignal component being a corresponding residual signal; the methodcomprising the acts of: determining a set of filter parameters of aprediction filter such that the prediction filter provides an estimateof the second signal component when receiving the first signal componentas an input; controlling the prediction filter by an error signalindicative of a difference of the second signal component and theestimate of the second signal component; representing the multi-channelsignal as the first signal component and the set of filter parameters;and transforming at least first and second source signal components ofthe multi-channel source signal by a predetermined transformation intothe principal component signal including most of the signal energy andat least the residual signal including less energy than the principalcomponent signal, the predetermined transformation being parameterizedby at least one transformation parameter; wherein the act ofrepresenting the multi-channel signal as the first signal component andthe set of filter parameters further comprises the act of representingthe multi-channel signal as the principal component signal, the set offilter parameters, and the at least one transformation parameter.
 2. Themethod according to claim 1, wherein the act of determining the set offilter parameters comprises the act of determining the filter parameterssuch that the error signal is smaller than a predetermined value.
 3. Themethod according to claim 1, wherein the act of representing themulti-channel signal as the first signal component and the set of filterparameters further comprises the act of representing the multi-channelsignal as the first signal component, the set of filter parameters, andthe error signal if the error signal is not smaller than a predeterminedvalue.
 4. The method according to claim 1, further comprising the act oftransforming at least the first and second source signal components ofthe multi-channel source signal into the first and second signalcomponents.
 5. The method according to claim 1, wherein themulti-channel source signal comprises a stereophonic signal including aleft signal component and a right signal component.
 6. A method ofencoding a multi channel signal including at least a first signalcomponent and a second signal component, the method comprising the actsof: determining a set of filter parameters of a prediction filter suchthat the prediction filter provides an estimate of the second signalcomponent when receiving the first signal component as an input; andrepresenting the multi channel signal as the first signal component andthe set of filter parameters; wherein said first signal component is aprincipal component signal of a source multi-channel signal including anumber of source signal components and the second signal component is acorresponding residual signal; the method further comprises the act oftransforming at least the first and second source signal components by apredetermined transformation into the principal component signalincluding most of the signal energy and at least the residual signalincluding less energy than the principal component signal, thepredetermined transformation being parameterized by at least onetransformation parameter; and the act of representing the multi-channelsignal as the first signal component and the set of filter parametersfurther comprises the act of representing the multi-channel signal asthe principal component signal, the set of filter parameters, and the atleast one transformation parameter.
 7. The method according to claim 6,wherein the predetermined transformation is a rotation and the at leastone transformation parameter corresponds to an angle of rotation.
 8. Amethod of encoding a multi channel signal including at least a firstsignal component and a second signal component, the first signalcomponent being a principal component signal of a multi-channel sourcesignal including a number of source signal components and the secondsignal component being a corresponding residual signal; the methodcomprising the acts of: determining a set of filter parameters of aprediction filter such that the prediction filter provides an estimateof the second signal component when receiving the first signal componentas an input; representing the multi channel signal as the first signalcomponent and the set of filter parameters; and transforming at leastfirst and second source signal components of the multi-channel sourcesignal by a predetermined transformation into the principal componentsignal including most of the signal energy and at least the residualsignal including less energy than the principal component signal, thepredetermined transformation being parameterized by at least onetransformation parameter; wherein the act of representing themulti-channel signal as the first signal component and the set of filterparameters further comprises the act of representing the multi-channelsignal as the principal component signal, the set of filter parameters,and the at least one transformation parameter, and wherein the act ofdetermining a set of filter parameters further comprises the act ofdetermining at least one scaling parameter for scaling the estimate ofthe second signal component such that a measure of correlation betweenthe second signal component and the estimate of the second signalcomponent is increased.
 9. A method of decoding multi-channel signalinformation, the method comprising the acts of: receiving a first signalcomponent and a set of filter parameters of an adaptive filtercontrolled by an error signal indicative of a difference of a secondsignal component and an estimate of the second signal component, whereinthe act of receiving the first signal component further comprises theact of receiving at least one transformation parameter, the first signalcomponent corresponding to a result of a predetermined transformation ofat least a first source signal component and a second source signalcomponent of a source multi-channel signal, the predeterminedtransformation being parameterized by the at least one transformationparameter; estimating the second signal component using a predictionfilter corresponding to the received set of filter parameters of theadaptive filter, the prediction filter receiving the received firstsignal component as an input; and generating a first decoded signalcomponent and a second decoded signal component by inverselytransforming the received first signal component and the estimatedsecond signal component.
 10. A method of decoding multi-channel signalinformation, the method comprising the acts of: receiving a first signalcomponent and a set of filter parameters; and estimating a second signalcomponent using a prediction filter corresponding to the received set offilter parameters, the prediction filter receiving the received firstsignal component as an input; wherein the act of receiving the firstsignal component further comprises the act of receiving a transformationparameter, the first signal component corresponding to a result of apredetermined transformation of at least a first source signal componentand a second source signal component of a source multi-channel signal,the predetermined transformation being parameterized by at least thetransformation parameter; and the method further comprises the act ofgenerating a first decoded signal component and a second decoded signalcomponent by inversely transforming the received first signal componentand the estimated second signal component.
 11. An arrangement forencoding a multi-channel signal including at least a first signalcomponent and a second signal component, the first signal componentbeing a principal component signal of a multi-channel source signalincluding a number of source signal components and the second signalcomponent being a corresponding residual signal; the arrangementcomprising: a prediction filter for estimating the second signalcomponent, the prediction filter corresponding to a set of filterparameters and receiving the first signal component as an input, whereinthe prediction filter is controlled by an error signal indicative of adifference of the second signal component and an estimate of the secondsignal component; and a processor configured for representing themulti-channel signal as the first signal component and the set of filterparameters including representing the multi-channel signal as theprincipal component signal, the set of filter parameters, and the atleast one transformation parameter; the processor being furtherconfigured for transforming at least the first and second source signalcomponents by a predetermined transformation into the principalcomponent signal including most of the signal energy and at least theresidual signal including less energy than the principal componentsignal, the predetermined transformation being parameterized by at leastone transformation parameter.
 12. An arrangement for decoding amulti-channel signal corresponding to at least two signal components,the arrangement comprising: receiving means for receiving a first signalcomponent of the multi-channel signal, a set of filter parameters of anadaptive filter controlled by an error signal indicative of a differenceof a second signal component and an estimate of the second signalcomponent, and at least one transformation parameter, the first signalcomponent corresponding to a result of a predetermined transformation ofat least a first source signal component and a second source signalcomponent of a source multi-channel signal, the predeterminedtransformation being parameterized by the at least one transformationparameter; a prediction filter for estimating the second signalcomponent of the multichannel signal, the prediction filter receivingthe received set of filter parameters of the adaptive filter and thereceived first signal component as an input; and a decoder configured togenerate a first decoded signal component and a second decoded signalcomponent by inversely transforming the first signal component and theestimated second signal component.
 13. A data signal includingmulti-channel signal information, the data signal being generated by amethod of encoding a multi-channel signal including at least a firstsignal component and a second signal component, the first signalcomponent being a principal component signal of a multi-channel sourcesignal including a number of source signal components and the secondsignal component being a corresponding residual signal; the methodcomprising the acts of: determining a set of filter parameters of aprediction filter such that the prediction filter provides an estimateof the second signal component when receiving the first signal componentas an input; controlling the prediction filter by an error signalindicative of a difference of the second signal component and theestimate of the second signal component; representing the multi-channelsignal as the first signal component and the set of filter parameters;and transforming at least first and second source signal components ofthe multi-channel source signal by a predetermined transformation intothe principal component signal including most of the signal energy andat least the residual signal including less energy than the principalcomponent signal, the predetermined transformation being parameterizedby at least one transformation parameter; wherein the act ofrepresenting the multi-channel signal as the first signal component andthe set of filter parameters further comprises the act of representingthe multi-channel signal as the principal component signal, the set offilter parameters, and the at least one transformation parameter.
 14. Acomputer-readable medium comprising a data record indicative ofmulti-channel signal information generated by a method of encoding amulti-channel signal including at least a first signal component and asecond signal component, the first signal component being a principalcomponent signal of a multi-channel source signal including a number ofsource signal components and the second signal component being acorresponding residual signal; the method comprising the acts of:determining a set of filter parameters of a prediction filter such thatthe prediction filter provides an estimate of the second signalcomponent when receiving the first signal component as an input;controlling the prediction filter by an error signal indicative of adifference of the second signal component and the estimate of the secondsignal component; representing the multi-channel signal as the firstsignal component and the set of filter parameters; and transforming atleast first and second source signal components of the multi-channelsource signal by a predetermined transformation into the principalcomponent signal including most of the signal energy and at least theresidual signal including less energy than the principal componentsignal, the predetermined transformation being parameterized by at leastone transformation parameter; wherein the act of representing themulti-channel signal as the first signal component and the set of filterparameters further comprises the act of representing the multi-channelsignal as the principal component signal, the set of filter parameters,and the at least one transformation parameter.
 15. A device forcommunicating a multi-channel signal, the device comprising anarrangement for encoding a multi-channel signal including at least afirst signal component and a second signal component, the first signalcomponent being a principal component signal of a multi-channel sourcesignal including a number of source signal components and the secondsignal component being a corresponding residual signal; the arrangementcomprising: a prediction filter for estimating the second signalcomponent, the prediction filter corresponding to a set of filterparameters and receiving the first signal component as an input, whereinthe prediction filter is controlled by an error signal indicative of adifference of the second signal component and an estimate of the secondsignal component; and a processor configured for representing themultichannel signal as the first signal component and the set of filterparameters including representing the multi-channel signal as theprincipal component signal, the set of filter parameters, and the atleast one transformation parameter; the processor being furtherconfigured for transforming at least the first and second source signalcomponents by a predetermined transformation into the principalcomponent signal including most of the signal energy and at least theresidual signal including less energy than the principal componentsignal, the predetermined transformation being parameterized by at leastone transformation parameter.